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- Bad call from PSTN to 9012XX1XX1
- Call is in Queue waiting for agent to pick up
- Asterisk18*CLI> core show channels
- Channel Location State Application(Data)
- SIP/9013XX9XX8-00000 9012XX1XX1@irock.com Up Queue(irock.com,t)
- 1 active channel
- 1 active call
- 32 calls processed
- Asterisk18*CLI> sip show peers
- Name/username Host Dyn Forcerport ACL Port Status Realtime
- 173.xx.xx.63 173.xx.xx.63 5060 Unmonitored
- 9012xx2xx4/9012xx2xx4 173.xx.xx.107 N 5060 Unmonitored Cached RT
- 9013XX9XX8/9013XX9XX8 173.xx.xx.107 N 5060 Unmonitored Cached RT
- 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
- -- Told SIP/9013XX9XX8-0000002b in irock.com their queue position (which was 1)
- -- <SIP/9013XX9XX8-0000002b> Playing 'queue-thankyou.gsm' (language 'en')
- <--- SIP read from UDP:173.xx.xx.107:5060 --->
- INVITE sip:9012XX1XX1@irock.com SIP/2.0
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
- To: <sip:9012XX1XX1@irock.com>
- CSeq: 103 INVITE
- Call-ID: B2B.365.2114915
- Content-Length: 332
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
- Content-Type: application/sdp
- Supported: replaces
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Max-Forwards: 68
- Contact: <sip:173.203.82.88:5060;transport=udp>
- P-hint: outbound->inbound
- P-hint: Route[6]: mediaproxy
- v=0
- o=root 15753 15753 IN IP4 64.2.142.15
- s=session
- c=IN IP4 64.2.142.15
- t=0 0
- m=audio 16632 RTP/AVP 0 8 3 18 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (17 headers 16 lines) ---
- Sending to 173.xx.xx.107:5060 (no NAT)
- Using INVITE request as basis request - B2B.365.2114915
- Found peer '9013XX9XX8' for '9014817029' from 173.xx.xx.107:5060
- == 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 18
- 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 G729 for ID 18
- Found audio description format telephone-event for ID 101
- Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 64.2.142.15:16632
- Looking for 9012XX1XX1 in irock.com (domain irock.com)
- list_route: hop: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- <--- Transmitting (NAT) to 173.xx.xx.107:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
- To: <sip:9012XX1XX1@irock.com>
- Call-ID: B2B.365.2114915
- CSeq: 103 INVITE
- Server: Asterisk PBX 1.8.4.2
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:9012XX1XX1@173.xx.xx.63:5060>
- Content-Length: 0
- <------------>
- -- Executing [9012XX1XX1@irock.com:1] Answer("SIP/9013XX9XX8-0000002c", "") in new stack
- Audio is at 5060
- Adding codec 0x2 (gsm) to SDP
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (NAT) to 173.xx.xx.107:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
- To: <sip:9012XX1XX1@irock.com>;tag=as588736bf
- Call-ID: B2B.365.2114915
- CSeq: 103 INVITE
- Server: Asterisk PBX 1.8.4.2
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:9012XX1XX1@173.xx.xx.63:5060>
- Content-Type: application/sdp
- Content-Length: 312
- v=0
- o=root 1750099629 1750099629 IN IP4 173.xx.xx.63
- s=Asterisk PBX 1.8.4.2
- c=IN IP4 173.xx.xx.63
- t=0 0
- m=audio 11978 RTP/AVP 3 0 8 101
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------>
- <--- SIP read from UDP:173.xx.xx.107:5060 --->
- ACK sip:9012XX1XX1@173.xx.xx.63:5060 SIP/2.0
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b>
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.2
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0
- To: <sip:9012XX1XX1@irock.com>;tag=as588736bf
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
- CSeq: 103 ACK
- Call-ID: B2B.365.2114915
- Content-Length: 0
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
- Max-Forwards: 69
- Contact: <sip:173.203.82.88:5060;transport=udp>
- <------------->
- --- (12 headers 0 lines) ---
- -- Executing [9012XX1XX1@irock.com:2] Set("SIP/9013XX9XX8-0000002c", "QUEUE_MAX_PENALTY=10") in new stack
- -- Executing [9012XX1XX1@irock.com:3] Queue("SIP/9013XX9XX8-0000002c", "irock.com,t") in new stack
- -- Started music on hold, class 'default', on SIP/9013XX9XX8-0000002c
- -- Stopped music on hold on SIP/9013XX9XX8-0000002c
- -- <SIP/9013XX9XX8-0000002c> Playing 'queue-youarenext.gsm' (language 'en')
- U 2011/06/14 16:03:48.757522 173.203.82.88:5060 -> 173.xx.xx.107:5060
- INVITE sip:9012XX1XX1@173.xx.xx.107:5060 SIP/2.0.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <9012XX1XX1@173.xx.xx.107>.
- CSeq: 103 INVITE.
- Call-ID: B2B.365.2114915.
- Content-Length: 332.
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
- Content-Type: application/sdp.
- Supported: replaces.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
- Max-Forwards: 69.
- Contact: <sip:173.203.82.88:5060;transport=udp>.
- .
- v=0.
- o=root 15753 15753 IN IP4 64.2.142.15.
- s=session.
- c=IN IP4 64.2.142.15.
- t=0 0.
- m=audio 16632 RTP/AVP 0 8 3 18 101.
- a=rtpmap:0 PCMU/8000.
- a=rtpmap:8 PCMA/8000.
- a=rtpmap:3 GSM/8000.
- a=rtpmap:18 G729/8000.
- a=fmtp:18 annexb=no.
- a=rtpmap:101 telephone-event/8000.
- a=fmtp:101 0-16.
- a=silenceSupp:off - - - -.
- a=ptime:20.
- a=sendrecv.
- .
- #
- U 2011/06/14 16:03:49.758654 173.203.82.88:5060 -> 173.xx.xx.107:5060
- INVITE sip:9012XX1XX1@173.xx.xx.107:5060 SIP/2.0.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <9012XX1XX1@173.xx.xx.107>.
- CSeq: 103 INVITE.
- Call-ID: B2B.365.2114915.
- Content-Length: 332.
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
- Content-Type: application/sdp.
- Supported: replaces.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
- Max-Forwards: 69.
- Contact: <sip:173.203.82.88:5060;transport=udp>.
- .
- v=0.
- o=root 15753 15753 IN IP4 64.2.142.15.
- s=session.
- c=IN IP4 64.2.142.15.
- t=0 0.
- m=audio 16632 RTP/AVP 0 8 3 18 101.
- a=rtpmap:0 PCMU/8000.
- a=rtpmap:8 PCMA/8000.
- a=rtpmap:3 GSM/8000.
- a=rtpmap:18 G729/8000.
- a=fmtp:18 annexb=no.
- a=rtpmap:101 telephone-event/8000.
- a=fmtp:101 0-16.
- a=silenceSupp:off - - - -.
- a=ptime:20.
- a=sendrecv.
- .
- #
- U 2011/06/14 16:03:50.315548 173.xx.xx.107:5060 -> 173.203.82.88:5060
- SIP/2.0 100 Giving a try.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <9012XX1XX1@173.xx.xx.107>.
- CSeq: 103 INVITE.
- Call-ID: B2B.365.2114915.
- Server: ae SIP Proxy.
- Content-Length: 0.
- .
- #
- U 2011/06/14 16:03:50.315709 173.xx.xx.107:5060 -> 173.xx.xx.63:5060
- INVITE sip:9012XX1XX1@irock.com SIP/2.0.
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- .
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <sip:9012XX1XX1@irock.com>.
- CSeq: 103 INVITE.
- Call-ID: B2B.365.2114915.
- Content-Length: 332.
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
- Content-Type: application/sdp.
- Supported: replaces.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
- Max-Forwards: 68.
- Contact: <sip:173.203.82.88:5060;transport=udp>.
- P-hint: outbound->inbound .
- P-hint: Route[6]: mediaproxy .
- .
- v=0.
- o=root 15753 15753 IN IP4 64.2.142.15.
- s=session.
- c=IN IP4 64.2.142.15.
- t=0 0.
- m=audio 16632 RTP/AVP 0 8 3 18 101.
- a=rtpmap:0 PCMU/8000.
- a=rtpmap:8 PCMA/8000.
- a=rtpmap:3 GSM/8000.
- a=rtpmap:18 G729/8000.
- a=fmtp:18 annexb=no.
- a=rtpmap:101 telephone-event/8000.
- a=fmtp:101 0-16.
- a=silenceSupp:off - - - -.
- a=ptime:20.
- a=sendrecv.
- #
- U 2011/06/14 16:03:50.316504 173.xx.xx.107:5060 -> 173.203.82.88:5060
- SIP/2.0 100 Giving a try.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <9012XX1XX1@173.xx.xx.107>.
- CSeq: 103 INVITE.
- Call-ID: B2B.365.2114915.
- Server: ae SIP Proxy.
- Content-Length: 0.
- .
- #
- U 2011/06/14 16:03:50.318765 173.xx.xx.63:5060 -> 173.xx.xx.107:5060
- SIP/2.0 100 Trying.
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- .
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <sip:9012XX1XX1@irock.com>.
- Call-ID: B2B.365.2114915.
- CSeq: 103 INVITE.
- Server: Asterisk PBX 1.8.4.2.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
- Supported: replaces, timer.
- Contact: <sip:9012XX1XX1@173.xx.xx.63:5060>.
- Content-Length: 0.
- .
- #
- U 2011/06/14 16:03:50.319325 173.xx.xx.63:5060 -> 173.xx.xx.107:5060
- SIP/2.0 200 OK.
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- .
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <sip:9012XX1XX1@irock.com>;tag=as588736bf.
- Call-ID: B2B.365.2114915.
- CSeq: 103 INVITE.
- Server: Asterisk PBX 1.8.4.2.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
- Supported: replaces, timer.
- Contact: <sip:9012XX1XX1@173.xx.xx.63:5060>.
- Content-Type: application/sdp.
- Content-Length: 312.
- .
- v=0.
- o=root 1750099629 1750099629 IN IP4 173.xx.xx.63.
- s=Asterisk PBX 1.8.4.2.
- c=IN IP4 173.xx.xx.63.
- t=0 0.
- m=audio 11978 RTP/AVP 3 0 8 101.
- a=rtpmap:3 GSM/8000.
- a=rtpmap:0 PCMU/8000.
- a=rtpmap:8 PCMA/8000.
- a=rtpmap:101 telephone-event/8000.
- a=fmtp:101 0-16.
- a=silenceSupp:off - - - -.
- a=ptime:20.
- a=sendrecv.
- #
- U 2011/06/14 16:03:50.551048 173.xx.xx.107:5060 -> 173.203.82.88:5060
- SIP/2.0 200 OK.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
- .
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- To: <9012XX1XX1@173.xx.xx.107>;tag=as588736bf.
- Call-ID: B2B.365.2114915.
- CSeq: 103 INVITE.
- Server: Asterisk PBX 1.8.4.2.
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
- Supported: replaces, timer.
- Contact: <sip:9012XX1XX1@173.xx.xx.63:5060>.
- Content-Type: application/sdp.
- Content-Length: 312.
- .
- v=0.
- o=root 1750099629 1750099629 IN IP4 173.xx.xx.63.
- s=Asterisk PBX 1.8.4.2.
- c=IN IP4 173.xx.xx.63.
- t=0 0.
- m=audio 11978 RTP/AVP 3 0 8 101.
- a=rtpmap:3 GSM/8000.
- a=rtpmap:0 PCMU/8000.
- a=rtpmap:8 PCMA/8000.
- a=rtpmap:101 telephone-event/8000.
- a=fmtp:101 0-16.
- a=silenceSupp:off - - - -.
- a=ptime:20.
- a=sendrecv.
- #
- U 2011/06/14 16:03:50.586584 173.203.82.88:5060 -> 173.xx.xx.107:5060
- ACK sip:9012XX1XX1@173.xx.xx.63:5060 SIP/2.0.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0.
- To: <9012XX1XX1@173.xx.xx.107>;tag=as588736bf.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- CSeq: 103 ACK.
- Call-ID: B2B.365.2114915.
- Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->.
- Content-Length: 0.
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
- Max-Forwards: 70.
- Contact: <sip:173.203.82.88:5060;transport=udp>.
- .
- .
- #
- U 2011/06/14 16:03:50.635278 173.xx.xx.107:5060 -> 173.xx.xx.63:5060
- ACK sip:9012XX1XX1@173.xx.xx.63:5060 SIP/2.0.
- Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b>.
- Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.2.
- Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0.
- To: <sip:9012XX1XX1@irock.com>;tag=as588736bf.
- From: <sip:9014817029@64.2.142.15>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
- CSeq: 103 ACK.
- Call-ID: B2B.365.2114915.
- Content-Length: 0.
- User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
- Max-Forwards: 69.
- Contact: <sip:173.203.82.88:5060;transport=udp>.
- .
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