Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- <--- SIP read from UDP:95.26.67.247:5062 --->
- INVITE sip:79250287897@sip2.bad-times.wtf SIP/2.0
- Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK498352626
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Contact: <sip:root@95.26.67.247:5062>
- Max-Forwards: 70
- User-Agent: qutecom/rev-g-trunk
- Expires: 120
- Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
- Content-Type: application/sdp
- Content-Length: 370
- v=0
- o=userX 20000001 20000001 IN IP4 95.26.67.247
- s=A call
- c=IN IP4 95.26.67.247
- t=1335435751 1335439351
- m=audio 10600 RTP/AVP 0 8 3 9 101
- a=rtpmap:0 PCMU/8000/1
- a=rtpmap:8 PCMA/8000/1
- a=rtpmap:3 GSM/8000/1
- a=rtpmap:9 G722/8000/1
- a=rtpmap:101 telephone-event/8000/1
- a=ptime:20
- m=video 10702 RTP/AVP 34 31
- a=rtpmap:34 H263/90000/1
- a=rtpmap:31 H261/90000/1
- <------------->
- --- (13 headers 15 lines) ---
- Sending to 95.26.67.247:5062 (NAT)
- Using INVITE request as basis request - 1621070911@192.168.0.25
- Found peer 'root' for 'root' from 95.26.67.247:5062
- == Using SIP RTP CoS mark 5
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 3
- Found RTP audio format 9
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found audio description format GSM for ID 3
- Found audio description format G722 for ID 9
- Found audio description format telephone-event for ID 101
- Found RTP video format 34
- Found RTP video format 31
- Found video description format H263 for ID 34
- Found video description format H261 for ID 31
- Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|g722)/video=(h261|h263)/text=(nothing), combined - (ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 95.26.67.247:10600
- Looking for 79250287897 in sip (domain sip2.bad-times.wtf)
- list_route: hop: <sip:root@95.26.67.247:5062>
- <--- Transmitting (NAT) to 95.26.67.247:5062 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Length: 0
- <------------>
- -- Executing [79250287897@sip:1] Dial("SIP/root-00000004", "SIP/skypeost/79250287897") in new stack
- == Using SIP RTP CoS mark 5
- -- Called SIP/skypeost/79250287897
- -- SIP/skypeost-00000005 is ringing
- <--- Transmitting (NAT) to 95.26.67.247:5062 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Length: 0
- <------------>
- -- SIP/skypeost-00000005 answered SIP/root-00000004
- Audio is at 18214
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100004 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (NAT) to 95.26.67.247:5062 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- <------------>
- -- Locally bridging SIP/root-00000004 and SIP/skypeost-00000005
- Retransmitting #1 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- Retransmitting #2 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- <--- SIP read from UDP:95.26.67.247:5062 --->
- OPTIONS sip:root@sip2.bad-times.wtf SIP/2.0
- Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK1534818002
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=1936133642
- To: <sip:root@sip2.bad-times.wtf>
- Call-ID: 393535928@192.168.0.25
- CSeq: 20 OPTIONS
- Max-Forwards: 70
- User-Agent: qutecom/rev-g-trunk
- Expires: 120
- Accept: application/sdp
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Looking for root in default (domain sip2.bad-times.wtf)
- <--- Transmitting (NAT) to 95.26.67.247:5062 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1534818002;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=1936133642
- To: <sip:root@sip2.bad-times.wtf>;tag=as4bbf5af1
- Call-ID: 393535928@192.168.0.25
- CSeq: 20 OPTIONS
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Accept: application/sdp
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '393535928@192.168.0.25' in 32000 ms (Method: OPTIONS)
- Retransmitting #3 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- Really destroying SIP dialog '1287725240@192.168.0.25' Method: OPTIONS
- Retransmitting #4 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- Retransmitting #5 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- Retransmitting #6 (NAT) to 95.26.67.247:5062:
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
- From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- Call-ID: 1621070911@192.168.0.25
- CSeq: 20 INVITE
- Server: Asterisk PBX 10.3.1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:79250287897@22.22.22.22:5060>
- Content-Type: application/sdp
- Content-Length: 284
- v=0
- o=root 226525189 226525189 IN IP4 22.22.22.22
- s=Asterisk PBX 10.3.1
- c=IN IP4 22.22.22.22
- t=0 0
- m=audio 18214 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34 31
- ---
- [Apr 26 14:22:33] WARNING[3194]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 1621070911@192.168.0.25 for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
- Packet timed out after 6400ms with no response
- [Apr 26 14:22:33] WARNING[3194]: chan_sip.c:3692 retrans_pkt: Hanging up call 1621070911@192.168.0.25 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
- == Spawn extension (sip, 79250287897, 1) exited non-zero on 'SIP/root-00000004'
- Scheduling destruction of SIP dialog '1621070911@192.168.0.25' in 6400 ms (Method: INVITE)
- set_destination: Parsing <sip:root@95.26.67.247:5062> for address/port to send to
- set_destination: set destination to 95.26.67.247:5062
- Reliably Transmitting (NAT) to 95.26.67.247:5062:
- BYE sip:root@95.26.67.247:5062 SIP/2.0
- Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport
- Max-Forwards: 70
- From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- Call-ID: 1621070911@192.168.0.25
- CSeq: 102 BYE
- User-Agent: Asterisk PBX 10.3.1
- X-Asterisk-HangupCause: Protocol error, unspecified
- X-Asterisk-HangupCauseCode: 111
- Content-Length: 0
- ---
- <--- SIP read from UDP:95.26.67.247:5062 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport=5060
- From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- Call-ID: 1621070911@192.168.0.25
- CSeq: 102 BYE
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:95.26.67.247:5062 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport=5060
- From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
- To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
- Call-ID: 1621070911@192.168.0.25
- CSeq: 102 BYE
- Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- SIP Response message for INCOMING dialog BYE arrived
- Really destroying SIP dialog '1621070911@192.168.0.25' Method: INVITE
- ster*CLI>
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement