12 May
2015
12 May
'15
11:06 a.m.
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...",..@..5....g.v......k.bINVITE sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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..."-..@.......g.v.......g.ACK sip:PHONENUMBER@PROVIDERIP SIP/2.0
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bK6ff2.2194a8f3123aacc04a451656d6e2f11a.0
Max-Forwards: 69
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=815f2ea990888c6d5eab0fa409f04ec4.44f3
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 102 ACK
Content-Length: 0
17:28:46.176460 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 1332
E..P"...@..b....g.v......<.IINVITE sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
User-Agent: Elastix 3.0
Proxy-Authorization: Digest username="PROVIDERUSER",
realm="PROVIDERIP", algorithm=MD5, uri="sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..)"/..@.......g.v........YACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v........jACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.......J.ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@.......g.v.........ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 103 INVITE
Server: Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:PHONENUMBER@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..@..~....g.v.......K.ACK sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@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@PROVIDERIP>;tag=as59947d90
To: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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@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...":..@.......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@PROVIDERIP>;tag=as59947d90
To: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as2db615d2
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@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.."";..@.......g.v.......I?BYE sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 104 BYE
User-Agent: Elastix 3.0
Proxy-Authorization: Digest username="PROVIDERUSER",
realm="PROVIDERIP", algorithm=MD5,
uri="sip:PHONENUMBER@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@PROVIDERIP:5080>;tag=as2db615d2
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as59947d90
Call-ID: 59e0216b4abc3a3d0c40eb2c33707ed7@PROVIDERIP
CSeq: 104 BYE
Server: Enswitch SIP proxy
Content-Length: 0
________________________________
From: sr-users [sr-users-bounces(a)lists.sip-router.org] on behalf of Daniel-Constantin
Mierla [miconda(a)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@PROVIDERIP:5080><sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP><sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@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@PROVIDERMEDIAIP:5060><sip:PHONENUMBER@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..@..v....g.v.......T.ACK<mailto:E..)!A..@..v....g.v.......T.ACK>
sip:PHONENUMBER@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@PROVIDERIP:5080><sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP><sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Contact: <sip:PROVIDERUSER@127.0.0.1:5080><sip:PROVIDERUSER@127.0.0.1:5080>^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M
CSeq: 103 ACK^M
User-Agent: Elastix 3.0^M
Content-Length: 0^M
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
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