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- <--- SIP read from UDP:10.224.253.172:60606 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK7d7ff5e0;rport
- From: "asterisk" <sip:[email protected]>;tag=as472e58e6
- To: <sip:10.10.10.1>;tag=14479C0C-183B
- Date: Fri, 18 Mar 2022 10:27:22 GMT
- Call-ID: [email protected]:5060
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- CSeq: 102 OPTIONS
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Accept: application/sdp
- Supported: 100rel,timer,resource-priority,replaces,sdp-anat
- Content-Type: application/sdp
- Content-Length: 381
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 5009 2983 IN IP4 10.224.253.172
- s=SIP Call
- c=IN IP4 10.224.253.172
- t=0 0
- m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
- c=IN IP4 10.224.253.172
- m=image 0 udptl t38
- c=IN IP4 10.224.253.172
- a=T38FaxVersion:0
- a=T38MaxBitRate:9600
- a=T38FaxRateManagement:transferredTCF
- a=T38FaxMaxBuffer:200
- a=T38FaxMaxDatagram:320
- a=T38FaxUdpEC:t38UDPRedundancy
- <------------->
- --- (14 headers 15 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.2:44702 --->
- INVITE sip:[email protected];transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6f03c555e61b405c;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP>
- To: <sip:[email protected]>
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 1 INVITE
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Content-Type: application/sdp
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 258
- v=0
- o=Z 4000401 1 IN IP4 10.10.10.2
- s=Z
- c=IN IP4 10.10.10.2
- t=0 0
- m=audio 41062 RTP/AVPF 8 0 102 9 106
- a=rtpmap:102 G726-32/8000
- a=rtpmap:106 opus/48000/2
- a=fmtp:106 minptime=20; useinbandfec=1
- a=sendrecv
- a=rtcp-fb:* nack pli
- a=rtcp-fb:* ccm fir
- <------------->
- --- (13 headers 12 lines) ---
- Sending to 10.10.10.2:44702 (NAT)
- Sending to 10.10.10.2:44702 (NAT)
- Using INVITE request as basis request - PJlX-9vhcAsgQ-rEp-uLRg..
- Found peer '7001' for '7001' from 10.10.10.2:44702
- == Using SIP RTP CoS mark 5
- [Mar 18 11:19:26] NOTICE[501004][C-0000000f]: chan_sip.c:10472 process_sdp: Received AVPF profile in audio offer but AVPF is not enabled, enabling: audio 41062 RTP/AVPF 8 0 102 9 106
- Found RTP audio format 8
- Found RTP audio format 0
- Found RTP audio format 102
- Found RTP audio format 9
- Found RTP audio format 106
- Found audio description format G726-32 for ID 102
- Found audio description format opus for ID 106
- Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g722|g726|opus)/video=(nothing)/text=(nothing), combined - (alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
- > 0x7f095c032180 -- Strict RTP learning after remote address set to: 10.10.10.2:41062
- Peer audio RTP is at port 10.10.10.2:41062
- Looking for 58675777 in internal (domain 10.10.10.2)
- sip_route_dump: route/path hop: <sip:[email protected]:44702;transport=UDP>
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6f03c555e61b405c;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- To: <sip:[email protected]>
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 1 INVITE
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Contact: <sip:[email protected]:5060>
- Content-Length: 0
- <------------>
- -- Executing [58675777@internal:1] NoOp("SIP/7001-0000001c", "Tunisia number is being dialed") in new stack
- -- Executing [58675777@internal:2] Set("SIP/7001-0000001c", "CALLERID(num)=39143961") in new stack
- -- Executing [58675777@internal:3] Dial("SIP/7001-0000001c", "SIP/58675777@trunk,45,To") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 11786
- Adding codec alaw to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 10.10.10.1:5060:
- INVITE sip:[email protected] SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK37d21ef5;rport
- Max-Forwards: 70
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:19:26 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 259
- v=0
- o=root 671092856 671092856 IN IP4 10.10.10.2
- s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
- c=IN IP4 10.10.10.2
- t=0 0
- m=audio 11786 RTP/AVP 8 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=maxptime:150
- a=sendrecv
- ---
- -- Called SIP/58675777@trunk
- <--- SIP read from UDP:10.10.10.1:64977 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK37d21ef5;rport
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>
- Date: Fri, 18 Mar 2022 10:27:29 GMT
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- Allow-Events: telephone-event
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- Session-ID: 00000000000000000000000000000000;remote=03ef7feb4f355b28baf82703f466599d
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- <--- SIP read from UDP:10.10.10.1:64977 --->
- OPTIONS sip:10.10.10.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF12199
- From: <sip:10.10.10.1>;tag=1447BE18-C42
- To: <sip:10.10.10.2>
- Date: Fri, 18 Mar 2022 10:27:31 GMT
- Call-ID: [email protected]
- User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
- Max-Forwards: 70
- CSeq: 101 OPTIONS
- Contact: <sip:10.10.10.1:5060>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to 10.10.10.1:64977 (NAT)
- Looking for s in internal (domain 10.10.10.2)
- <--- Transmitting (NAT) to 10.10.10.1:64977 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF12199;received=10.10.10.1;rport=64977
- From: <sip:10.10.10.1>;tag=1447BE18-C42
- To: <sip:10.10.10.2>;tag=as5c802cb7
- Call-ID: [email protected]
- CSeq: 101 OPTIONS
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
- <--- SIP read from UDP:10.10.10.1:64977 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK37d21ef5;rport
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>;tag=1447C054-CF1
- Date: Fri, 18 Mar 2022 10:27:29 GMT
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Contact: <sip:[email protected]:5060>
- Supported: sdp-anat
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- Session-ID: 4a931221a95c5d41b94f2852f52a4313;remote=03ef7feb4f355b28baf82703f466599d
- Content-Type: application/sdp
- Content-Disposition: session;handling=required
- Content-Length: 241
- P-Early-Media: sendrecv
- P-Early-Media: sendrecv
- P-Early-Media: sendrecv
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 9955 5508 IN IP4 10.10.10.1
- s=SIP Call
- c=IN IP4 10.10.10.1
- t=0 0
- m=audio 17080 RTP/AVP 8 101
- c=IN IP4 10.10.10.1
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- <------------->
- --- (19 headers 11 lines) ---
- sip_route_dump: route/path hop: <sip:[email protected]:5060>
- Found RTP audio format 8
- Found RTP audio format 101
- Found audio description format PCMA for ID 8
- Found audio description format telephone-event for ID 101
- Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- > 0x7f0980005410 -- Strict RTP learning after remote address set to: 10.10.10.1:17080
- Peer audio RTP is at port 10.10.10.1:17080
- -- SIP/trunk-0000001d is making progress passing it to SIP/7001-0000001c
- Audio is at 10884
- Adding codec alaw to SDP
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6f03c555e61b405c;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- To: <sip:[email protected]>;tag=as750a6146
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 1 INVITE
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Contact: <sip:[email protected]:5060>
- Content-Type: application/sdp
- Content-Length: 206
- v=0
- o=root 1518744979 1518744979 IN IP4 10.10.10.2
- s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
- c=IN IP4 10.10.10.2
- t=0 0
- m=audio 10884 RTP/AVPF 8
- a=rtpmap:8 PCMA/8000
- a=ptime:20
- a=maxptime:150
- a=sendrecv
- <------------>
- > 0x7f095c032180 -- Strict RTP qualifying stream type: audio
- > 0x7f095c032180 -- Strict RTP switching source address to 10.10.10.2:58644
- <--- SIP read from UDP:10.10.10.1:64977 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK37d21ef5;rport
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>;tag=1447C054-CF1
- Date: Fri, 18 Mar 2022 10:27:29 GMT
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
- Contact: <sip:[email protected]:5060>
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- Session-ID: 4a931221a95c5d41b94f2852f52a4313;remote=03ef7feb4f355b28baf82703f466599d
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- sip_route_dump: route/path hop: <sip:[email protected]:5060>
- -- SIP/trunk-0000001d is ringing
- [Mar 18 11:19:29] WARNING[519080][C-0000000f]: channel.c:4559 indicate_data_internal: Unable to handle indication 3 for 'SIP/7001-0000001c'
- > 0x7f0980005410 -- Strict RTP switching to RTP target address 10.10.10.1:17080 as source
- > 0x7f095c032180 -- Strict RTP learning complete - Locking on source address 10.10.10.2:58644
- <--- SIP read from UDP:10.10.10.2:44702 --->
- REGISTER sip:10.10.10.2;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6fe0508845e82f17;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- To: "Wamia prod"<sip:[email protected];transport=UDP>
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 37 REGISTER
- Expires: 60
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Sending to 10.10.10.2:44702 (NAT)
- Sending to 10.10.10.2:44702 (NAT)
- Reliably Transmitting (NAT) to 10.10.10.2:44702:
- OPTIONS sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK72c22a1f;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as5f205d43
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:19:32 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6fe0508845e82f17;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=as2e03c937
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 37 REGISTER
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Expires: 60
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;expires=60
- Date: Fri, 18 Mar 2022 10:19:32 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK72c22a1f;rport=5060
- Contact: <sip:10.0.2.15:36627>
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;tag=a6bbd808
- From: "asterisk" <sip:[email protected]>;tag=as5f205d43
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- Accept: application/sdp, application/sdp
- Accept-Language: en
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- > 0x7f0980005410 -- Strict RTP learning complete - Locking on source address 10.10.10.1:17080
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- <--- SIP read from UDP:10.10.10.1:64977 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK37d21ef5;rport
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>;tag=1447C054-CF1
- Date: Fri, 18 Mar 2022 10:27:29 GMT
- Call-ID: [email protected]:5060
- CSeq: 102 INVITE
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
- Contact: <sip:[email protected]:5060>
- Supported: replaces
- Supported: sdp-anat
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- Session-ID: 4a931221a95c5d41b94f2852f52a4313;remote=03ef7feb4f355b28baf82703f466599d
- Supported: timer
- Content-Type: application/sdp
- Content-Disposition: session;handling=required
- Content-Length: 241
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 9955 5508 IN IP4 10.10.10.1
- s=SIP Call
- c=IN IP4 10.10.10.1
- t=0 0
- m=audio 17080 RTP/AVP 8 101
- c=IN IP4 10.10.10.1
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- <------------->
- --- (19 headers 11 lines) ---
- sip_route_dump: route/path hop: <sip:[email protected]:5060>
- Transmitting (NAT) to 10.10.10.1:64977:
- ACK sip:[email protected]:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK361def19;rport
- Max-Forwards: 70
- From: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- To: <sip:[email protected]>;tag=1447C054-CF1
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Content-Length: 0
- ---
- -- SIP/trunk-0000001d answered SIP/7001-0000001c
- Audio is at 10884
- Adding codec alaw to SDP
- <--- Reliably Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---6f03c555e61b405c;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- To: <sip:[email protected]>;tag=as750a6146
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 1 INVITE
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Contact: <sip:[email protected]:5060>
- Content-Type: application/sdp
- Content-Length: 206
- v=0
- o=root 1518744979 1518744979 IN IP4 10.10.10.2
- s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
- c=IN IP4 10.10.10.2
- t=0 0
- m=audio 10884 RTP/AVPF 8
- a=rtpmap:8 PCMA/8000
- a=ptime:20
- a=maxptime:150
- a=sendrecv
- <------------>
- -- Channel SIP/trunk-0000001d joined 'simple_bridge' basic-bridge <33255246-8dad-4d75-94a7-86bd7e1302d5>
- -- Channel SIP/7001-0000001c joined 'simple_bridge' basic-bridge <33255246-8dad-4d75-94a7-86bd7e1302d5>
- <--- SIP read from UDP:10.10.10.2:44702 --->
- ACK sip:[email protected]:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---35cf3aa73dfa76a7;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP>
- To: <sip:[email protected]>;tag=as750a6146
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 1 ACK
- User-Agent: Z 5.5.10 v2.10.17.3
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]' Method: OPTIONS
- Really destroying SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' Method: REGISTER
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- Reliably Transmitting (NAT) to 10.10.10.1:5060:
- OPTIONS sip:10.10.10.1 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK1fe1ab83;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as577cfa32
- To: <sip:10.10.10.1>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:20:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- SIP read from UDP:10.224.253.172:60606 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK1fe1ab83;rport
- From: "asterisk" <sip:[email protected]>;tag=as577cfa32
- To: <sip:10.10.10.1>;tag=1448866C-8D6
- Date: Fri, 18 Mar 2022 10:28:22 GMT
- Call-ID: [email protected]:5060
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- CSeq: 102 OPTIONS
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Accept: application/sdp
- Supported: 100rel,timer,resource-priority,replaces,sdp-anat
- Content-Type: application/sdp
- Content-Length: 381
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 3167 5767 IN IP4 10.224.253.172
- s=SIP Call
- c=IN IP4 10.224.253.172
- t=0 0
- m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
- c=IN IP4 10.224.253.172
- m=image 0 udptl t38
- c=IN IP4 10.224.253.172
- a=T38FaxVersion:0
- a=T38MaxBitRate:9600
- a=T38FaxRateManagement:transferredTCF
- a=T38FaxMaxBuffer:200
- a=T38FaxMaxDatagram:320
- a=T38FaxUdpEC:t38UDPRedundancy
- <------------->
- --- (14 headers 15 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.2:44702 --->
- REGISTER sip:10.10.10.2;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---d2874ebe62aa4913;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- To: "Wamia prod"<sip:[email protected];transport=UDP>
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 38 REGISTER
- Expires: 60
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Sending to 10.10.10.2:44702 (NAT)
- Sending to 10.10.10.2:44702 (NAT)
- Reliably Transmitting (NAT) to 10.10.10.2:44702:
- OPTIONS sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK3895bb30;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as58857db7
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:20:26 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---d2874ebe62aa4913;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=as31a2e193
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 38 REGISTER
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Expires: 60
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;expires=60
- Date: Fri, 18 Mar 2022 10:20:26 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK3895bb30;rport=5060
- Contact: <sip:10.0.2.15:36627>
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;tag=ac97c36f
- From: "asterisk" <sip:[email protected]>;tag=as58857db7
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- Accept: application/sdp, application/sdp
- Accept-Language: en
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.1:64977 --->
- OPTIONS sip:10.10.10.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF5376
- From: <sip:10.10.10.1>;tag=1448A87C-262B
- To: <sip:10.10.10.2>
- Date: Fri, 18 Mar 2022 10:28:31 GMT
- Call-ID: [email protected]
- User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
- Max-Forwards: 70
- CSeq: 101 OPTIONS
- Contact: <sip:10.10.10.1:5060>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to 10.10.10.1:64977 (NAT)
- Looking for s in internal (domain 10.10.10.2)
- <--- Transmitting (NAT) to 10.10.10.1:64977 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF5376;received=10.10.10.1;rport=64977
- From: <sip:10.10.10.1>;tag=1448A87C-262B
- To: <sip:10.10.10.2>;tag=as698679d9
- Call-ID: [email protected]
- CSeq: 101 OPTIONS
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- Really destroying SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' Method: REGISTER
- Really destroying SIP dialog '[email protected]' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- Reliably Transmitting (NAT) to 10.10.10.1:5060:
- OPTIONS sip:10.10.10.1 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK5b63e61d;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as0791bc7d
- To: <sip:10.10.10.1>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:21:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- SIP read from UDP:10.224.253.172:60606 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK5b63e61d;rport
- From: "asterisk" <sip:[email protected]>;tag=as0791bc7d
- To: <sip:10.10.10.1>;tag=144970C8-C3B
- Date: Fri, 18 Mar 2022 10:29:22 GMT
- Call-ID: [email protected]:5060
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- CSeq: 102 OPTIONS
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Accept: application/sdp
- Supported: 100rel,timer,resource-priority,replaces,sdp-anat
- Content-Type: application/sdp
- Content-Length: 381
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 3088 1161 IN IP4 10.224.253.172
- s=SIP Call
- c=IN IP4 10.224.253.172
- t=0 0
- m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
- c=IN IP4 10.224.253.172
- m=image 0 udptl t38
- c=IN IP4 10.224.253.172
- a=T38FaxVersion:0
- a=T38MaxBitRate:9600
- a=T38FaxRateManagement:transferredTCF
- a=T38FaxMaxBuffer:200
- a=T38FaxMaxDatagram:320
- a=T38FaxUdpEC:t38UDPRedundancy
- <------------->
- --- (14 headers 15 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.2:44702 --->
- REGISTER sip:10.10.10.2;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---b3542810db6b5d7c;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- To: "Wamia prod"<sip:[email protected];transport=UDP>
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 39 REGISTER
- Expires: 60
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Sending to 10.10.10.2:44702 (NAT)
- Sending to 10.10.10.2:44702 (NAT)
- Reliably Transmitting (NAT) to 10.10.10.2:44702:
- OPTIONS sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK287567c7;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as2bb9925a
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:21:20 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---b3542810db6b5d7c;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=as4ebd3331
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 39 REGISTER
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Expires: 60
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;expires=60
- Date: Fri, 18 Mar 2022 10:21:20 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK287567c7;rport=5060
- Contact: <sip:10.0.2.15:36627>
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;tag=78a06a67
- From: "asterisk" <sip:[email protected]>;tag=as2bb9925a
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- Accept: application/sdp, application/sdp
- Accept-Language: en
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.1:64977 --->
- OPTIONS sip:10.10.10.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF71394
- From: <sip:10.10.10.1>;tag=144992DC-2150
- To: <sip:10.10.10.2>
- Date: Fri, 18 Mar 2022 10:29:31 GMT
- Call-ID: [email protected]
- User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
- Max-Forwards: 70
- CSeq: 101 OPTIONS
- Contact: <sip:10.10.10.1:5060>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to 10.10.10.1:64977 (NAT)
- Looking for s in internal (domain 10.10.10.2)
- <--- Transmitting (NAT) to 10.10.10.1:64977 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF71394;received=10.10.10.1;rport=64977
- From: <sip:10.10.10.1>;tag=144992DC-2150
- To: <sip:10.10.10.2>;tag=as112de737
- Call-ID: [email protected]
- CSeq: 101 OPTIONS
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- Really destroying SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' Method: REGISTER
- Really destroying SIP dialog '[email protected]' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.2:44702 --->
- <------------->
- <--- SIP read from UDP:10.10.10.2:44702 --->
- REGISTER sip:10.10.10.2;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---db1a2b85e3173b74;rport
- Max-Forwards: 70
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- To: "Wamia prod"<sip:[email protected];transport=UDP>
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 40 REGISTER
- Expires: 60
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Sending to 10.10.10.2:44702 (NAT)
- Sending to 10.10.10.2:44702 (NAT)
- Reliably Transmitting (NAT) to 10.10.10.2:44702:
- OPTIONS sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK4284a3fd;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as1b915abe
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:22:14 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- Transmitting (NAT) to 10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.2.15:36627;branch=z9hG4bK-524287-1---db1a2b85e3173b74;received=10.10.10.2;rport=44702
- From: "Wamia prod"<sip:[email protected];transport=UDP>;tag=0aa60b15
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=as7b52b653
- Call-ID: dyhRQvAT0TlGzHOHHW3b-Q..
- CSeq: 40 REGISTER
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Expires: 60
- Contact: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;expires=60
- Date: Fri, 18 Mar 2022 10:22:14 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'dyhRQvAT0TlGzHOHHW3b-Q..' in 32000 ms (Method: REGISTER)
- <--- SIP read from UDP:10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK4284a3fd;rport=5060
- Contact: <sip:10.0.2.15:36627>
- To: <sip:[email protected]:44702;transport=UDP;rinstance=3582acae9ce8adbb>;tag=f8c4363b
- From: "asterisk" <sip:[email protected]>;tag=as1b915abe
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- Accept: application/sdp, application/sdp
- Accept-Language: en
- Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
- Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
- User-Agent: Z 5.5.10 v2.10.17.3
- Allow-Events: presence, kpml, talk
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- Reliably Transmitting (NAT) to 10.10.10.1:5060:
- OPTIONS sip:10.10.10.1 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK431b8a84;rport
- Max-Forwards: 70
- From: "asterisk" <sip:[email protected]>;tag=as70b056c1
- To: <sip:10.10.10.1>
- Contact: <sip:[email protected]:5060>
- Call-ID: [email protected]:5060
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Date: Fri, 18 Mar 2022 10:22:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- ---
- <--- SIP read from UDP:10.224.253.172:60606 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK431b8a84;rport
- From: "asterisk" <sip:[email protected]>;tag=as70b056c1
- To: <sip:10.10.10.1>;tag=144A5B2C-1F01
- Date: Fri, 18 Mar 2022 10:30:22 GMT
- Call-ID: [email protected]:5060
- Server: Cisco-SIPGateway/IOS-15.7.3.M
- CSeq: 102 OPTIONS
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
- Allow-Events: telephone-event
- Accept: application/sdp
- Supported: 100rel,timer,resource-priority,replaces,sdp-anat
- Content-Type: application/sdp
- Content-Length: 381
- v=0
- o=CiscoSystemsSIP-GW-UserAgent 2186 4146 IN IP4 10.224.253.172
- s=SIP Call
- c=IN IP4 10.224.253.172
- t=0 0
- m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
- c=IN IP4 10.224.253.172
- m=image 0 udptl t38
- c=IN IP4 10.224.253.172
- a=T38FaxVersion:0
- a=T38MaxBitRate:9600
- a=T38FaxRateManagement:transferredTCF
- a=T38FaxMaxBuffer:200
- a=T38FaxMaxDatagram:320
- a=T38FaxUdpEC:t38UDPRedundancy
- <------------->
- --- (14 headers 15 lines) ---
- Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- <--- SIP read from UDP:10.10.10.1:64977 --->
- OPTIONS sip:10.10.10.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF922B8
- From: <sip:10.10.10.1>;tag=144A7D40-253C
- To: <sip:10.10.10.2>
- Date: Fri, 18 Mar 2022 10:30:31 GMT
- Call-ID: [email protected]
- User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
- Max-Forwards: 70
- CSeq: 101 OPTIONS
- Contact: <sip:10.10.10.1:5060>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to 10.10.10.1:64977 (NAT)
- Looking for s in internal (domain 10.10.10.2)
- <--- Transmitting (NAT) to 10.10.10.1:64977 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FF922B8;received=10.10.10.1;rport=64977
- From: <sip:10.10.10.1>;tag=144A7D40-253C
- To: <sip:10.10.10.2>;tag=as471d4f86
- Call-ID: [email protected]
- CSeq: 101 OPTIONS
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
- <--- SIP read from UDP:10.10.10.1:64977 --->
- BYE sip:[email protected]:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FFA19DE
- From: <sip:[email protected]>;tag=1447C054-CF1
- To: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- Date: Fri, 18 Mar 2022 10:27:41 GMT
- Call-ID: [email protected]:5060
- User-Agent: Cisco-SIPGateway/IOS-15.7.3.M
- Max-Forwards: 70
- Timestamp: 1647599432
- CSeq: 101 BYE
- Reason: Q.850;cause=16
- P-RTP-Stat: PS=8926,OS=1428160,PR=9001,OR=1440160,PL=0,JI=0,LA=0,DU=170
- Session-ID: 4a931221a95c5d41b94f2852f52a4313;remote=03ef7feb4f355b28baf82703f466599d
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Sending to 10.10.10.1:64977 (NAT)
- Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: BYE)
- <--- Transmitting (NAT) to 10.10.10.1:64977 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1FFA19DE;received=10.10.10.1;rport=64977
- From: <sip:[email protected]>;tag=1447C054-CF1
- To: "Wamia prod" <sip:[email protected]>;tag=as725a98d6
- Call-ID: [email protected]:5060
- CSeq: 101 BYE
- Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces
- Content-Length: 0
- <------------>
- -- Channel SIP/trunk-0000001d left 'simple_bridge' basic-bridge <33255246-8dad-4d75-94a7-86bd7e1302d5>
- -- Channel SIP/7001-0000001c left 'simple_bridge' basic-bridge <33255246-8dad-4d75-94a7-86bd7e1302d5>
- == Spawn extension (internal, 58675777, 3) exited non-zero on 'SIP/7001-0000001c'
- Scheduling destruction of SIP dialog 'PJlX-9vhcAsgQ-rEp-uLRg..' in 6400 ms (Method: ACK)
- Reliably Transmitting (NAT) to 10.10.10.2:44702:
- BYE sip:[email protected]:44702;transport=UDP SIP/2.0
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK1a63028c;rport
- Max-Forwards: 70
- From: <sip:[email protected]>;tag=as750a6146
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 102 BYE
- User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- <--- SIP read from UDP:10.10.10.2:44702 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK1a63028c;rport=5060
- Contact: <sip:[email protected]:44702;transport=UDP>
- To: "Wamia prod"<sip:[email protected];transport=UDP>;tag=cb0ba13a
- From: <sip:[email protected]>;tag=as750a6146
- Call-ID: PJlX-9vhcAsgQ-rEp-uLRg..
- CSeq: 102 BYE
- User-Agent: Z 5.5.10 v2.10.17.3
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- SIP Response message for INCOMING dialog BYE arrived
- Really destroying SIP dialog 'PJlX-9vhcAsgQ-rEp-uLRg..' Method: ACK
- Really destroying SIP dialog '[email protected]:5060' Method: BYE
- <--- SIP read from UDP:10.10.10.2:44702 --->
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