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<div style="direction: ltr;font-family: Tahoma;color: #000000;font-size: 10pt;">Here's the full conversation. Makes me wonder whether the ACK needs to go back to the same host that handled the INVITE or whether it should be returned to the host mentioned in
"c=IN IP4 PROVIDERMEDIAIP" in the 200 OK.<br>
<br>
<br>
<br>
17:28:46.129459 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 1123<br>
E...",..@..5....g.v......k.bINVITE sip:PHONENUMBER@PROVIDERIP SIP/2.0<br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7384433b;rport=5080<br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP><br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 102 INVITE<br>
User-Agent: Elastix 3.0<br>
Date: Tue, 12 May 2015 07:28:46 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 301<br>
P-hint: outbound<br>
<br>
v=0<br>
o=root 2142344521 2142344521 IN IP4 172.21.0.226<br>
s=Asterisk PBX 11.13.0<br>
c=IN IP4 172.21.0.226<br>
t=0 0<br>
m=audio 19840 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
a=nortpproxy:yes<br>
<br>
17:28:46.170220 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 566<br>
E..R.0..?..^g.v..........>.3SIP/2.0 407 Proxy Authentication Required<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0;rport=5060<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7384433b;rport=5080<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 102 INVITE<br>
Proxy-Authenticate: Digest realm="PROVIDERIP", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN"<br>
Server: Enswitch SIP proxy<br>
Content-Length: 0<br>
<br>
<br>
17:28:46.170606 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 382<br>
E..."-..@.......g.v.......g.ACK sip:PHONENUMBER@PROVIDERIP SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0<br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 102 ACK<br>
Content-Length: 0<br>
<br>
<br>
17:28:46.176460 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 1332<br>
E..P"...@..b....g.v......<.IINVITE sip:PHONENUMBER@PROVIDERIP SIP/2.0<br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP><br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
User-Agent: Elastix 3.0<br>
Proxy-Authorization: Digest username="PROVIDERUSER", realm="PROVIDERIP", algorithm=MD5, uri="sip:PHONENUMBER@PROVIDERIP", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN", response="75ea690ebdd7bfa9eabf0e9f2c298bcc"<br>
Date: Tue, 12 May 2015 07:28:46 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 301<br>
P-hint: outbound<br>
<br>
v=0<br>
o=root 2142344521 2142344522 IN IP4 172.21.0.226<br>
s=Asterisk PBX 11.13.0<br>
c=IN IP4 172.21.0.226<br>
t=0 0<br>
m=audio 19840 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
a=nortpproxy:yes<br>
<br>
17:28:46.219802 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 441<br>
E....1..?...g.v.............SIP/2.0 100 trying -- your call is important to us<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0;rport=5060<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch SIP proxy<br>
Content-Length: 0<br>
<br>
<br>
17:28:52.359718 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1070<br>
E..J.2..?.<br>
dg.v..........6b.SIP/2.0 183 Session Progress<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494236 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:28:54.281615 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.3..?.<br>
qg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:28:54.286312 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"/..@.......g.v........YACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.63b0410c520626648931a7b1cf931791.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK22240b78;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:28:54.781431 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.4..?.<br>
pg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:28:54.784927 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"0..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.d59e40b6e47afc80a1daf9b4e2803373.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK08fceb3e;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:28:55.781287 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.5..?.<br>
og.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:28:55.786000 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"1..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.e6c93dc8958d6bf30d85cde34ecfb130.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK1752e724;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:28:57.780918 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.6..?.<br>
ng.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:28:57.784319 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"2..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.496a633f0de916ea0147b3323e426860.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK75465f90;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:01.780730 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.7..?.<br>
mg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:01.783005 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"3..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.4d1857b7912373c5e7e8041b4b249bc2.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2df438ae;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:05.781325 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.8..?.<br>
lg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:05.783799 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"4..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.750e819c84323da35eef87e564268658.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK32b5b7cd;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:09.780783 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.9..?.<br>
kg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:09.783343 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"5..@.......g.v........jACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.7f0e2010772d1442152c2444955d1155.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7ccbfc48;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:13.781533 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.:..?.<br>
jg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:13.784128 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"6..@.......g.v.......J.ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.235a361d94585070c1da6b5980c0ea3c.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK08d73c33;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:17.780975 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.;..?.<br>
ig.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:17.783305 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"7..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.16cdedbeb3c4a84877e1f9a60d53e3ea.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK63cd594c;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:21.780775 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.<..?.<br>
hg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:21.783062 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"8..@.......g.v.........ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.1fac9edf46f0eb1ea39cae8e09ae8189.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK5b6013cc;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:25.781427 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056<br>
E..<.=..?.<br>
gg.v..........(X.SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080<br>
Record-Route: <sip:PROVIDERIP;lr=on><br>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 INVITE<br>
Server: Enswitch<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces<br>
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060><br>
Content-Type: application/sdp<br>
Content-Length: 284<br>
<br>
v=0<br>
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP<br>
s=Asterisk PBX 11.3.0<br>
c=IN IP4 PROVIDERMEDIAIP<br>
t=0 0<br>
m=audio 19208 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
17:29:25.783614 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
E..)"9..@..~....g.v.......K.ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.98fb129b0c4c8e2b1c77a3a69dd97de4.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4048140d;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 103 ACK<br>
User-Agent: Elastix 3.0<br>
Content-Length: 0<br>
<br>
<br>
17:29:27.408747 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 706<br>
E....>..?...g.v............zBYE sip:PROVIDERUSER@172.21.0.226:5060 SIP/2.0<br>
Via: SIP/2.0/UDP PROVIDERIP;branch=z9hG4bK6ff2.ca437ba5.0<br>
Via: SIP/2.0/UDP PROVIDERMEDIAIP:5060;branch=z9hG4bK1236ad79;rport=5060<br>
Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes><br>
Max-Forwards: 69<br>
From: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
To: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 102 BYE<br>
User-Agent: Enswitch<br>
X-Asterisk-HangupCause: No user responding<br>
X-Asterisk-HangupCauseCode: 18<br>
Content-Length: 0<br>
<b>X-Enswitch-RURI: sip:PROVIDERUSER@172.21.0.226:5060<br>
X-Enswitch-Source: PROVIDERMEDIAIP:5060</b><br>
<br>
<br>
17:29:27.412081 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 418<br>
E...":..@.......g.v.........SIP/2.0 404 Not here<br>
Via: SIP/2.0/UDP PROVIDERIP;rport=5060;branch=z9hG4bK6ff2.ca437ba5.0<br>
Via: SIP/2.0/UDP PROVIDERMEDIAIP:5060;branch=z9hG4bK1236ad79;rport=5060<br>
From: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
To: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 102 BYE<br>
Server: kamailio (4.1.6 (x86_64/linux))<br>
Content-Length: 0<br>
<br>
<br>
17:29:41.468388 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 774<br>
E.."";..@.......g.v.......I?BYE sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK4ff2.b971e9b05b7c9fbbb5e63fd94973e216.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK73b4aae5;rport=5080<br>
Route: <sip:PROVIDERIP;lr=on><br>
Max-Forwards: 69<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 104 BYE<br>
User-Agent: Elastix 3.0<br>
Proxy-Authorization: Digest username="PROVIDERUSER", realm="PROVIDERIP", algorithm=MD5, uri="sip:PHONENUMBER@PROVIDERMEDIAIP:5060", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN", response="49aab6f0725de9b6c146f92d64b26b8a"<br>
X-Asterisk-HangupCause: Normal Clearing<br>
X-Asterisk-HangupCauseCode: 16<br>
Content-Length: 0<br>
<br>
<br>
17:29:41.506107 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 424<br>
E....?..?...g.v............[SIP/2.0 404 Not found<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK4ff2.b971e9b05b7c9fbbb5e63fd94973e216.0<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK73b4aae5;rport=5080<br>
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2<br>
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90<br>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP<br>
CSeq: 104 BYE<br>
Server: Enswitch SIP proxy<br>
Content-Length: 0<br>
<br>
<br>
<br>
<div style="font-family: Times New Roman; color: #000000; font-size: 16px">
<hr tabindex="-1">
<div style="direction: ltr;" id="divRpF297748"><font color="#000000" face="Tahoma" size="2"><b>From:</b> sr-users [sr-users-bounces@lists.sip-router.org] on behalf of Daniel-Constantin Mierla [miconda@gmail.com]<br>
<b>Sent:</b> Tuesday, 12 May 2015 5:45 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Repeated 200 OK from Enswitch<br>
</font><br>
</div>
<div></div>
<div>Hello,<br>
<br>
can you show both received 200ok + ACK as well as those sent out? It is important to see how Record-/Route, Contact and r-uri change on the way to spot where the issue is.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 12/05/15 05:56, Darren Campbell (Primar) wrote:<br>
</div>
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<div style="direction:ltr; font-family:Tahoma; color:#000000; font-size:10pt">Hi all<br>
<br>
Experiencing a commonly reported issue where calls drop out after 30 seconds or so. Mainly because the provider hangs up after not recognising/receiving ACK in response to 200 OK.<br>
<br>
Unfortunately (or maybe fortunately), I haven't had much experience with Enswitch so was hoping someone in the community might help guide me as to which rules Enswitch might be using to match ACKs to calls in progress. Maybe there is another avenue I should
be investigating.<br>
<br>
<br>
Here's a sample of the 200 OK and ACK that repeats.<br>
<br>
13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1058<br>
E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M<br>
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERIP;lr=on" target="_blank">
<sip:PROVIDERIP;lr=on></a>^M<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes" target="_blank">
<sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes></a>^M<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes" target="_blank">
<sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes></a>^M<br>
From: "asterisk" <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@PROVIDERIP:5080" target="_blank">
<sip:PROVIDERUSER@PROVIDERIP:5080></a>;tag=as65919d92^M<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERIP" target="_blank">
<sip:PHONENUMBER@PROVIDERIP></a>;tag=as260fefaa^M<br>
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M<br>
CSeq: 103 INVITE^M<br>
Server: Enswitch^M<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M<br>
Supported: replaces^M<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERMEDIAIP:5060" target="_blank">
<sip:PHONENUMBER@PROVIDERMEDIAIP:5060></a>^M<br>
Content-Type: application/sdp^M<br>
Content-Length: 286^M<br>
^M<br>
v=0^M<br>
o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M<br>
s=Asterisk PBX 11.3.0^M<br>
c=IN IP4 PROVIDERMEDIAIP^M<br>
t=0 0^M<br>
m=audio 15594 RTP/AVP 0 8 3 101^M<br>
a=rtpmap:0 PCMU/8000^M<br>
a=rtpmap:8 PCMA/8000^M<br>
a=rtpmap:3 GSM/8000^M<br>
a=rtpmap:101 telephone-event/8000^M<br>
a=fmtp:101 0-16^M<br>
a=ptime:20^M<br>
a=sendrecv^M<br>
<br>
13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525<br>
<a class="moz-txt-link-abbreviated" href="mailto:E..)!A..@..v....g.v.......T.ACK" target="_blank">E..)!A..@..v....g.v.......T.ACK</a>
<a class="moz-txt-link-freetext" href="sip:PHONENUMBER@PROVIDERIP:5060" target="_blank">
sip:PHONENUMBER@PROVIDERIP:5060</a> SIP/2.0^M<br>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M<br>
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M<br>
Route: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERIP;lr=on" target="_blank">
<sip:PROVIDERIP;lr=on></a>^M<br>
Max-Forwards: 69^M<br>
From: "asterisk" <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@PROVIDERIP:5080" target="_blank">
<sip:PROVIDERUSER@PROVIDERIP:5080></a>;tag=as65919d92^M<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:PHONENUMBER@PROVIDERIP" target="_blank">
<sip:PHONENUMBER@PROVIDERIP></a>;tag=as260fefaa^M<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:PROVIDERUSER@127.0.0.1:5080" target="_blank">
<sip:PROVIDERUSER@127.0.0.1:5080></a>^M<br>
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M<br>
CSeq: 103 ACK^M<br>
User-Agent: Elastix 3.0^M<br>
Content-Length: 0^M<br>
<br>
</div>
<br>
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<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - <a class="moz-txt-link-freetext" href="http://www.kamailioworld.com" target="_blank">http://www.kamailioworld.com</a></pre>
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