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^M Record-Route: 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^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: 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^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 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^M Max-Forwards: 69^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: sip:PHONENUMBER@PROVIDERIP;tag=as260fefaa^M Contact: 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
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^M Record-Route: 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^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: 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^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 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^M Max-Forwards: 69^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: sip:PHONENUMBER@PROVIDERIP;tag=as260fefaa^M Contact: 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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@lists.sip-router.org] on behalf of Daniel-Constantin Mierla [miconda@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=onsip:PROVIDERIP;lr=on^M Record-Route: sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yessip: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=yessip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: sip:PHONENUMBER@PROVIDERIPsip: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:5060sip: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.ACKmailto: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=onsip:PROVIDERIP;lr=on^M Max-Forwards: 69^M From: "asterisk" sip:PROVIDERUSER@PROVIDERIP:5080sip:PROVIDERUSER@PROVIDERIP:5080;tag=as65919d92^M To: sip:PHONENUMBER@PROVIDERIPsip:PHONENUMBER@PROVIDERIP;tag=as260fefaa^M Contact: sip:PROVIDERUSER@127.0.0.1:5080sip: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
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orgmailto: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
On 12/05/15 10:06, Darren Campbell (Primar) wrote:
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.
You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel
Thank you Daniel, is it OK to send these logs directly to you? I'd prefer to not disclose them to the list.
________________________________ From: Daniel-Constantin Mierla [miconda@gmail.com] Sent: Tuesday, 12 May 2015 7:03 PM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
On 12/05/15 10:06, Darren Campbell (Primar) wrote: 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. You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel
-- 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
Yes, you can send me directly, just notify here so I will look in the global inbox, because I check it quire rarely -- I is a lot of unsolicited emails I get there and I have filters for the mailing list, so if no filter is selecting the email to put it in a dedicated folder, it may take ages to look at it if I am not aware of being sent.
Cheers, Daniel
On 15/05/15 04:51, Darren Campbell (Primar) wrote:
Thank you Daniel, is it OK to send these logs directly to you? I'd prefer to not disclose them to the list.
*From:* Daniel-Constantin Mierla [miconda@gmail.com] *Sent:* Tuesday, 12 May 2015 7:03 PM *To:* Darren Campbell (Primar); Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] Repeated 200 OK from Enswitch
On 12/05/15 10:06, Darren Campbell (Primar) wrote:
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.
You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel -- 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
It appears the ACK had an R-URI from original INVITE and wasn't using the IP address in the new Contact from the 200 OK.
Commented out a call to fix_nated_sdp(); which resolved this case, now completing a round of testing to check whether any new faults have been introduced from the change.
________________________________ From: Daniel-Constantin Mierla [miconda@gmail.com] Sent: Saturday, 16 May 2015 3:57 AM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
Yes, you can send me directly, just notify here so I will look in the global inbox, because I check it quire rarely -- I is a lot of unsolicited emails I get there and I have filters for the mailing list, so if no filter is selecting the email to put it in a dedicated folder, it may take ages to look at it if I am not aware of being sent.
Cheers, Daniel
On 15/05/15 04:51, Darren Campbell (Primar) wrote: Thank you Daniel, is it OK to send these logs directly to you? I'd prefer to not disclose them to the list.
________________________________ From: Daniel-Constantin Mierla [miconda@gmail.commailto:miconda@gmail.com] Sent: Tuesday, 12 May 2015 7:03 PM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
On 12/05/15 10:06, Darren Campbell (Primar) wrote: 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. You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel
-- Daniel-Constantin Mierla http://twitter.com/#!/micondahttp://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
-- 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
Hello,
but fix_nated_sdp() is changing the SDP (body) not Contact header. Maybe you had fix_nated_contact(). If yes, you should use set_contact_alias() instead, along with handle_ruri_alias() -- see default kamailio.cfg for version 4.2 for a sample of how to use these two functions.
Cheers, Daniel
On 18/05/15 08:31, Darren Campbell (Primar) wrote:
It appears the ACK had an R-URI from original INVITE and wasn't using the IP address in the new Contact from the 200 OK.
Commented out a call to fix_nated_sdp(); which resolved this case, now completing a round of testing to check whether any new faults have been introduced from the change.
*From:* Daniel-Constantin Mierla [miconda@gmail.com] *Sent:* Saturday, 16 May 2015 3:57 AM *To:* Darren Campbell (Primar); Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] Repeated 200 OK from Enswitch
Yes, you can send me directly, just notify here so I will look in the global inbox, because I check it quire rarely -- I is a lot of unsolicited emails I get there and I have filters for the mailing list, so if no filter is selecting the email to put it in a dedicated folder, it may take ages to look at it if I am not aware of being sent.
Cheers, Daniel
On 15/05/15 04:51, Darren Campbell (Primar) wrote:
Thank you Daniel, is it OK to send these logs directly to you? I'd prefer to not disclose them to the list.
*From:* Daniel-Constantin Mierla [miconda@gmail.com] *Sent:* Tuesday, 12 May 2015 7:03 PM *To:* Darren Campbell (Primar); Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] Repeated 200 OK from Enswitch
On 12/05/15 10:06, Darren Campbell (Primar) wrote:
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.
You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel -- 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
-- 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
Yes, that's the one, actually it was fix_nated_contact() that was commented out.
Thanks for the pointer to set_contact_alias() and handle_ruri_alias(). I see them already in the cfg here so hopefully they've been used correctly. Tests will tell. ________________________________ From: Daniel-Constantin Mierla [miconda@gmail.com] Sent: Monday, 18 May 2015 4:59 PM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
Hello,
but fix_nated_sdp() is changing the SDP (body) not Contact header. Maybe you had fix_nated_contact(). If yes, you should use set_contact_alias() instead, along with handle_ruri_alias() -- see default kamailio.cfg for version 4.2 for a sample of how to use these two functions.
Cheers, Daniel
On 18/05/15 08:31, Darren Campbell (Primar) wrote: It appears the ACK had an R-URI from original INVITE and wasn't using the IP address in the new Contact from the 200 OK.
Commented out a call to fix_nated_sdp(); which resolved this case, now completing a round of testing to check whether any new faults have been introduced from the change.
________________________________ From: Daniel-Constantin Mierla [miconda@gmail.commailto:miconda@gmail.com] Sent: Saturday, 16 May 2015 3:57 AM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
Yes, you can send me directly, just notify here so I will look in the global inbox, because I check it quire rarely -- I is a lot of unsolicited emails I get there and I have filters for the mailing list, so if no filter is selecting the email to put it in a dedicated folder, it may take ages to look at it if I am not aware of being sent.
Cheers, Daniel
On 15/05/15 04:51, Darren Campbell (Primar) wrote: Thank you Daniel, is it OK to send these logs directly to you? I'd prefer to not disclose them to the list.
________________________________ From: Daniel-Constantin Mierla [miconda@gmail.commailto:miconda@gmail.com] Sent: Tuesday, 12 May 2015 7:03 PM To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Repeated 200 OK from Enswitch
On 12/05/15 10:06, Darren Campbell (Primar) wrote: 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. You still gave the conversation in one side, towards the provider. You need to get the traffic from caller side as well, which is listening on 127.0.0.1, but it is important to see what that side sends and receives. You have to grab the traffic on lo interface as well.
Cheers, Daniel
-- Daniel-Constantin Mierla http://twitter.com/#!/micondahttp://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
-- Daniel-Constantin Mierla http://twitter.com/#!/micondahttp://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
-- 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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Daniel-Constantin Mierla -
Darren Campbell (Primar)