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- <--- SIP read from UDP:74.209.242.21:5060 --->
- INVITE tel:30441212792250 SIP/2.0
- Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;rport
- Max-Forwards: 70
- From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
- To: <tel:30441212792250>
- Contact: <sip:74.209.242.21>
- CSeq: 101 INVITE
- Call-Id: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
- Route: <sip:111.222.333.444;lr>
- User-Agent: Cisco-SIPGateway/IOS-12.x
- Supported: 100rel,timer,resource-priority,replaces
- Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, UPDATE, REFER SUBSCRIBE, NOTIFY, INFO
- Allow-Events: telephone-event
- Content-Type: application/sdp
- Content-Length: 306
- v=0
- o=unknown 10839 10840 IN IP4 189.172.50.191
- s=SIP Call
- c=IN IP4 189.172.50.191
- t=0 0
- m=audio 11000 RTP/AVP 18 4 0 8 3 101
- a=rtpmap:18 G729/8000
- a=rtpmap:4 G723/8000
- 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=sendrecv
- <------------->
- --- (15 headers 14 lines) ---
- == Using UDPTL TOS bits 184
- == Using UDPTL CoS mark 5
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to On
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ - INVITE (No RTP)
- Sending to 74.209.242.21:5060 (NAT)
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:21905 handle_request_invite: Initializing initreq for method INVITE - callid ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
- Using INVITE request as basis request - ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
- No matching peer for '74.209.242.21' from '74.209.242.21:5060'
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:345 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xa539ce0'
- [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:483 ast_rtp_new: Allocated port 15664 for RTP instance '0xa539ce0'
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:354 ast_rtp_instance_new: RTP instance '0xa539ce0' is setup and ready to go
- [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:2394 ast_rtp_prop_set: Setup RTCP on RTP instance '0xa539ce0'
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4934 do_setnat: Setting NAT on RTP to On
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to On
- Found RTP audio format 18
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0xb7b92370
- Found RTP audio format 4
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on 0xb7b92370
- Found RTP audio format 0
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb7b92370
- Found RTP audio format 8
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb7b92370
- Found RTP audio format 3
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0xb7b92370
- Found RTP audio format 101
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb7b92370
- Found audio description format G729 for ID 18
- Found audio description format G723 for ID 4
- 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 telephone-event for ID 101
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb7b92370
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 3 on 0xb7b92370
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0xb7b92370
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0xb7b92370
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0xb7b92370
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb7b92370
- Capabilities: us - 0x5104 (ulaw|g729|g722|siren14), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa539ce0'
- Peer audio RTP is at port 189.172.50.191:11000
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:9036 process_sdp: Peer doesn't provide T.38 UDPTL
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:22053 handle_request_invite: Checking SIP call limits for device
- [2013-06-21 16:43:49] DEBUG[26527]: sip/reqresp_parser.c:83 parse_uri_full: No supported scheme found in 'tel:30441212792250' using the scheme[s] sip:,sips:
- [2013-06-21 16:43:49] WARNING[26527]: chan_sip.c:14820 get_destination: Not a SIP header ()?
- <--- Reliably Transmitting (NAT) to 74.209.242.21:5060 --->
- SIP/2.0 416 Unsupported URI scheme
- Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;received=74.209.242.21;rport=5060
- From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
- To: <tel:30441212792250>;tag=as4e56dc9e
- Call-ID: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
- CSeq: 101 INVITE
- Server: FPBX-2.11.0(1.8.7.2)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' in 32000 ms (Method: INVITE)
- <--- SIP read from UDP:74.209.242.21:5060 --->
- ACK tel:30441212792250 SIP/2.0
- Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;received=74.209.242.21;rport
- From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
- To: <tel:30441212792250>;tag=as4e56dc9e
- Call-Id: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
- CSeq: 101 ACK
- User-Agent: gR-LxuidiMCFoH-
- Max-Forwards: 70
- Allow: INVITE, ACK, CANCEL, BYE
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4011 __sip_ack: Stopping retransmission on 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' of Response 101: Match Found
- Really destroying SIP dialog 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' Method: ACK
- [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:293 instance_destructor: Destroyed RTP instance '0xa539ce0'
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