[SR-Users] Repeated 200 OK from Enswitch

Tue May 12 10:06:49 CEST 2015

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.

17:28:46.129459 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 1123
E...",.. at ..5....g.v......k.bINVITE sip:PHONENUMBER at PROVIDERIP SIP/2.0
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7384433b;rport=5080
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 102 INVITE
User-Agent: Elastix 3.0
Date: Tue, 12 May 2015 07:28:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301
P-hint: outbound

v=0
o=root 2142344521 2142344521 IN IP4 172.21.0.226
s=Asterisk PBX 11.13.0
c=IN IP4 172.21.0.226
t=0 0
m=audio 19840 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes

17:28:46.170220 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 566
E..R.0..?..^g.v..........>.3SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0;rport=5060
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7384433b;rport=5080
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="PROVIDERIP", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN"
Server: Enswitch SIP proxy
Content-Length: 0

17:28:46.170606 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 382
E..."-.. at .......g.v.......g.ACK sip:PHONENUMBER at PROVIDERIP SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 102 ACK
Content-Length: 0

17:28:46.176460 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 1332
E..P"... at ..b....g.v......<.IINVITE sip:PHONENUMBER at PROVIDERIP SIP/2.0
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
User-Agent: Elastix 3.0
Proxy-Authorization: Digest username="PROVIDERUSER", realm="PROVIDERIP", algorithm=MD5, uri="sip:PHONENUMBER at PROVIDERIP", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN", response="75ea690ebdd7bfa9eabf0e9f2c298bcc"
Date: Tue, 12 May 2015 07:28:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301
P-hint: outbound

v=0
o=root 2142344521 2142344522 IN IP4 172.21.0.226
s=Asterisk PBX 11.13.0
c=IN IP4 172.21.0.226
t=0 0
m=audio 19840 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes

17:28:46.219802 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 441
E....1..?...g.v.............SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0;rport=5060
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch SIP proxy
Content-Length: 0

17:28:52.359718 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1070
E..J.2..?.
dg.v..........6b.SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494236 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:28:54.281615 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.3..?.
qg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:28:54.286312 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"/.. at .......g.v........YACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.63b0410c520626648931a7b1cf931791.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK22240b78;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:28:54.781431 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.4..?.
pg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:28:54.784927 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"0.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.d59e40b6e47afc80a1daf9b4e2803373.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK08fceb3e;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:28:55.781287 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.5..?.
og.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:28:55.786000 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"1.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.e6c93dc8958d6bf30d85cde34ecfb130.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK1752e724;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:28:57.780918 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.6..?.
ng.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:28:57.784319 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"2.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.496a633f0de916ea0147b3323e426860.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK75465f90;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:01.780730 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.7..?.
mg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:01.783005 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"3.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.4d1857b7912373c5e7e8041b4b249bc2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2df438ae;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:05.781325 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.8..?.
lg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:05.783799 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"4.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.750e819c84323da35eef87e564268658.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK32b5b7cd;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:09.780783 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.9..?.
kg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:09.783343 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"5.. at .......g.v........jACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.7f0e2010772d1442152c2444955d1155.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK7ccbfc48;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:13.781533 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.:..?.
jg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:13.784128 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"6.. at .......g.v.......J.ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.235a361d94585070c1da6b5980c0ea3c.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK08d73c33;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:17.780975 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.;..?.
ig.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:17.783305 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"7.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.16cdedbeb3c4a84877e1f9a60d53e3ea.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK63cd594c;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:21.780775 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.<..?.
hg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:21.783062 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"8.. at .......g.v.........ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.1fac9edf46f0eb1ea39cae8e09ae8189.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK5b6013cc;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:25.781427 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1056
E..<.=..?.
gg.v..........(X.SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK7ff2.815a4eaaa71a2b504d0e053b565d5085.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK21ff4fac;rport=5080
Record-Route: <sip:PROVIDERIP;lr=on>
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as2db615d2;nat=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as2db615d2;nat=yes>
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 750494236 750494237 IN IP4 PROVIDERMEDIAIP
s=Asterisk PBX 11.3.0
c=IN IP4 PROVIDERMEDIAIP
t=0 0
m=audio 19208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:29:25.783614 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)"9.. at ..~....g.v.......K.ACK sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK7ff2.98fb129b0c4c8e2b1c77a3a69dd97de4.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4048140d;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 103 ACK
User-Agent: Elastix 3.0
Content-Length: 0

17:29:27.408747 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 706
E....>..?...g.v............zBYE sip:PROVIDERUSER at 172.21.0.226:5060 SIP/2.0
Via: SIP/2.0/UDP PROVIDERIP;branch=z9hG4bK6ff2.ca437ba5.0
Via: SIP/2.0/UDP PROVIDERMEDIAIP:5060;branch=z9hG4bK1236ad79;rport=5060
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>
Max-Forwards: 69
From: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
To: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 102 BYE
User-Agent: Enswitch
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
X-Enswitch-RURI: sip:PROVIDERUSER at 172.21.0.226:5060
X-Enswitch-Source: PROVIDERMEDIAIP:5060

17:29:27.412081 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 418
E...":.. at .......g.v.........SIP/2.0 404 Not here
Via: SIP/2.0/UDP PROVIDERIP;rport=5060;branch=z9hG4bK6ff2.ca437ba5.0
Via: SIP/2.0/UDP PROVIDERMEDIAIP:5060;branch=z9hG4bK1236ad79;rport=5060
From: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
To: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 102 BYE
Server: kamailio (4.1.6 (x86_64/linux))
Content-Length: 0

17:29:41.468388 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 774
E.."";.. at .......g.v.......I?BYE sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK4ff2.b971e9b05b7c9fbbb5e63fd94973e216.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK73b4aae5;rport=5080
Route: <sip:PROVIDERIP;lr=on>
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 104 BYE
User-Agent: Elastix 3.0
Proxy-Authorization: Digest username="PROVIDERUSER", realm="PROVIDERIP", algorithm=MD5, uri="sip:PHONENUMBER at PROVIDERMEDIAIP:5060", nonce="VVGs2VVRq620ayXnC7qlie1+Jfz14FtN", response="49aab6f0725de9b6c146f92d64b26b8a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

17:29:41.506107 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 424
E....?..?...g.v............[SIP/2.0 404 Not found
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bK4ff2.b971e9b05b7c9fbbb5e63fd94973e216.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK73b4aae5;rport=5080
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER at PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7 at PROVIDERIP
CSeq: 104 BYE
Server: Enswitch SIP proxy
Content-Length: 0

________________________________
From: sr-users [sr-users-bounces at lists.sip-router.org] on behalf of Daniel-Constantin Mierla [miconda at gmail.com]
Sent: Tuesday, 12 May 2015 5:45 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Repeated 200 OK from Enswitch

Hello,

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.

Cheers,
Daniel

On 12/05/15 05:56, Darren Campbell (Primar) wrote:
Hi all

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.

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.

Here's a sample of the 200 OK and ACK that repeats.

13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1058
E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M
Record-Route: <sip:PROVIDERIP;lr=on><sip:PROVIDERIP;lr=on>^M
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes><sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes><sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080><sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER at PROVIDERIP><sip:PHONENUMBER at PROVIDERIP>;tag=as260fefaa^M
Call-ID: 271ac7a174d613cd0b94504353733a2c at PROVIDERIP^M
CSeq: 103 INVITE^M
Server: Enswitch^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces^M
Contact: <sip:PHONENUMBER at PROVIDERMEDIAIP:5060><sip:PHONENUMBER at PROVIDERMEDIAIP:5060>^M
Content-Type: application/sdp^M
Content-Length: 286^M
^M
v=0^M
o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M
s=Asterisk PBX 11.3.0^M
c=IN IP4 PROVIDERMEDIAIP^M
t=0 0^M
m=audio 15594 RTP/AVP 0 8 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M

13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)!A.. at ..v....g.v.......T.ACK<mailto:E..)!A.. at ..v....g.v.......T.ACK> sip:PHONENUMBER at PROVIDERIP:5060 SIP/2.0^M
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M
Route: <sip:PROVIDERIP;lr=on><sip:PROVIDERIP;lr=on>^M
Max-Forwards: 69^M
From: "asterisk" <sip:PROVIDERUSER at PROVIDERIP:5080><sip:PROVIDERUSER at PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER at PROVIDERIP><sip:PHONENUMBER at PROVIDERIP>;tag=as260fefaa^M
Contact: <sip:PROVIDERUSER at 127.0.0.1:5080><sip:PROVIDERUSER at 127.0.0.1:5080>^M
Call-ID: 271ac7a174d613cd0b94504353733a2c at PROVIDERIP^M
CSeq: 103 ACK^M
User-Agent: Elastix 3.0^M
Content-Length: 0^M

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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