-- Executing Dial("SIP/70-ee64", "SIP/695199@voip.inode.at||rtT") in new stack We're at 81.223.57.205 port 12350 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) 12 headers, 10 lines Reliably Transmitting (NAT) to 62.99.171.21:5060: INVITE sip:695199@voip.inode.at SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK43c8f6c4;rport From: "70" ;tag=as100f59dc To: Contact: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 21 Mar 2005 15:12:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 209 v=0 o=root 23089 23089 IN IP4 81.223.57.205 s=session c=IN IP4 81.223.57.205 t=0 0 m=audio 12350 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Called 695199@voip.inode.at JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK43c8f6c4;rport=5060 From: "70" ;tag=as100f59dc To: ;tag=677451b7b5396553087fdcc12290d0dd.7457 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voip.inode.at", nonce="423ed71f8861d4f021fc69c6d9658dbe39545106" Server: Sip EXpress router (0.9.1 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Transmitting (NAT) to 62.99.171.21:5060: ACK sip:695199@voip.inode.at SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK43c8f6c4;rport From: "70" ;tag=as100f59dc To: ;tag=677451b7b5396553087fdcc12290d0dd.7457 Contact: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- We're at 81.223.57.205 port 12350 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Reliably Transmitting (NAT) to 62.99.171.21:5060: INVITE sip:695199@voip.inode.at SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport From: "70" ;tag=as100f59dc To: Contact: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="0316890841", realm="voip.inode.at", algorithm=MD5, uri="sip:695199@voip.inode.at", nonce="423ed71f8861d4f021fc69c6d9658dbe39545106", response="287342529b9c07510bb08e4f516e07ac", opaque="" Date: Mon, 21 Mar 2005 15:12:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 209 v=0 o=root 23089 23090 IN IP4 81.223.57.205 s=session c=IN IP4 81.223.57.205 t=0 0 m=audio 12350 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport=5060 From: "70" ;tag=as100f59dc To: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE Server: Sip EXpress router (0.9.1 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport=5060 From: "70" ;tag=as100f59dc To: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE Server: Sip EXpress router (0.9.1 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 183 Session Progress Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport=5060 Record-Route: Content-Length: 0 Contact: sip:020316695199@62.99.171.52:5060 User-Agent: SN2400 MxSF v3.2.7.36 --- (10 headers 0 lines)--- -- SIP/voip.inode.at-7e33 is making progress passing it to SIP/70-ee64 JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 180 Ringing Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport=5060 Record-Route: Content-Length: 143 Content-Type: application/sdp Contact: sip:020316695199@62.99.171.52:5060 User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136596 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.52 t=0 0 m=audio 4892 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv --- (11 headers 8 lines)--- -- SIP/voip.inode.at-7e33 is ringing JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 200 OK Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 INVITE From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK2b233394;rport=5060 Record-Route: Content-Length: 143 Content-Type: application/sdp Supported: replaces Contact: sip:020316695199@62.99.171.52:5060 User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136601 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.52 t=0 0 m=audio 4892 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv --- (12 headers 8 lines)--- Found RTP audio format 8 Peer audio RTP is at port 62.99.171.52:4892 Found description format PCMA Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 62.99.171.21, port 5060 Transmitting (NAT) to 62.99.171.21:5060: ACK sip:020316695199@62.99.171.52:5060 SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK06d5e5dc;rport Route: From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Contact: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/voip.inode.at-7e33 answered SIP/70-ee64 -- Attempting native bridge of SIP/70-ee64 and SIP/voip.inode.at-7e33 JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: INVITE sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKd0d8.d951dfe4.0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bKbf6ca860d Max-Forwards: 16 Content-Length: 122 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536766 INVITE Supported: timer Content-Type: application/sdp Contact: sip:020316695199@62.99.171.52:5060 Supported: replaces User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136603 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.21 t=0 0 m=image 36150 udptl t38 a=sendrecv --- (15 headers 7 lines)--- Using latest request as basis request Sending to 62.99.171.21 : 5060 (NAT) Mar 21 16:12:07 WARNING[23078]: chan_sip.c:2886 process_sdp: Unknown SDP media type in offer: image 36150 udptl t38 Transmitting (NAT) to 62.99.171.21:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKd0d8.d951dfe4.0;received=62.99.171.21;rport=5060 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bKbf6ca860d From: ;tag=de56240ac355625 To: "70" ;tag=as100f59dc Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536766 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: ontent-Length: 0 --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: ACK sip:0316890841@81.223.57.205 SIP/2.0 Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKd0d8.d951dfe4.0 From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 To: "70" ;tag=as100f59dc CSeq: 396536766 ACK Route: User-Agent: Sip EXpress router(0.9.1 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: INVITE sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKe0d8.1655c746.0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK48dab2199 Max-Forwards: 16 Content-Length: 144 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536767 INVITE Supported: timer Content-Type: application/sdp Contact: sip:020316695199@62.99.171.52:5060 Supported: replaces User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136604 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.21 t=0 0 m=audio 36152 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv --- (15 headers 8 lines)--- Using latest request as basis request Sending to 62.99.171.21 : 5060 (NAT) Found RTP audio format 8 Peer audio RTP is at port 62.99.171.21:36152 Found description format PCMA Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 81.223.57.205 port 12350 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Reliably Transmitting (NAT) to 62.99.171.21:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKe0d8.1655c746.0;received=62.99.171.21;rport=5060 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK48dab2199 Record-Route: From: ;tag=de56240ac355625 To: "70" ;tag=as100f59dc Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536767 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 23089 23091 IN IP4 81.223.57.205 s=session c=IN IP4 81.223.57.205 t=0 0 m=audio 12350 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: ACK sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK514d852c4 Max-Forwards: 16 Content-Length: 0 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536767 ACK Contact: sip:020316695199@62.99.171.52:5060 User-Agent: SN2400 MxSF v3.2.7.36 --- (12 headers 0 lines)--- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: INVITE sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKc1d8.886c8352.0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK31dc12c2f Max-Forwards: 16 Content-Length: 122 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536768 INVITE Supported: timer Content-Type: application/sdp Contact: sip:020316695199@62.99.171.52:5060 Supported: replaces User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136605 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.21 t=0 0 m=image 36150 udptl t38 a=sendrecv --- (15 headers 7 lines)--- Using latest request as basis request Sending to 62.99.171.21 : 5060 (NAT) Mar 21 16:12:10 WARNING[23078]: chan_sip.c:2886 process_sdp: Unknown SDP media type in offer: image 36150 udptl t38 Transmitting (NAT) to 62.99.171.21:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKc1d8.886c8352.0;received=62.99.171.21;rport=5060 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK31dc12c2f From: ;tag=de56240ac355625 To: "70" ;tag=as100f59dc Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536768 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: ontent-Length: 0 --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: ACK sip:0316890841@81.223.57.205 SIP/2.0 Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKc1d8.886c8352.0 From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 To: "70" ;tag=as100f59dc CSeq: 396536768 ACK Route: User-Agent: Sip EXpress router(0.9.1 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: INVITE sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKb1d8.b350bbd1.0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK6e853783d Max-Forwards: 16 Content-Length: 144 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536769 INVITE Supported: timer Content-Type: application/sdp Contact: sip:020316695199@62.99.171.52:5060 Supported: replaces User-Agent: SN2400 MxSF v3.2.7.36 v=0 o=MxSIP 0 136606 IN IP4 62.99.171.52 s=SIP Call c=IN IP4 62.99.171.21 t=0 0 m=audio 36152 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv --- (15 headers 8 lines)--- Using latest request as basis request Sending to 62.99.171.21 : 5060 (NAT) Found RTP audio format 8 Peer audio RTP is at port 62.99.171.21:36152 Found description format PCMA Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 81.223.57.205 port 12350 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Reliably Transmitting (NAT) to 62.99.171.21:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 62.99.171.21;branch=z9hG4bKb1d8.b350bbd1.0;received=62.99.171.21;rport=5060 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK6e853783d Record-Route: From: ;tag=de56240ac355625 To: "70" ;tag=as100f59dc Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536769 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 23089 23092 IN IP4 81.223.57.205 s=session c=IN IP4 81.223.57.205 t=0 0 m=audio 12350 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: ACK sip:0316890841@81.223.57.205 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 62.99.171.21;branch=0 Via: SIP/2.0/UDP 62.99.171.52:5060;branch=z9hG4bK7d411b868 Max-Forwards: 16 Content-Length: 0 To: "70" ;tag=as100f59dc From: ;tag=de56240ac355625 Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 396536769 ACK Contact: sip:020316695199@62.99.171.52:5060 User-Agent: SN2400 MxSF v3.2.7.36 --- (12 headers 0 lines)--- 11 headers, 0 lines Reliably Transmitting (no NAT) to 62.99.171.21:5060: OPTIONS sip:voip.inode.at SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK7de34723 From: "asterisk" ;tag=as0bffae18 To: Contact: Call-ID: 5551b9f324f6ab8e634c574c24e99dd7@81.223.57.205 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 21 Mar 2005 15:12:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 --- JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK7de34723 From: "asterisk" ;tag=as0bffae18 To: ;tag=677451b7b5396553087fdcc12290d0dd.a7d1 Call-ID: 5551b9f324f6ab8e634c574c24e99dd7@81.223.57.205 CSeq: 102 OPTIONS Proxy-Authenticate: Digest realm="voip.inode.at", nonce="423ed736b986b3f394cda0c300d6fd6654bcbd61" Server: Sip EXpress router (0.9.1 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '5551b9f324f6ab8e634c574c24e99dd7@81.223.57.205' set_destination: Parsing for address/port to send to set_destination: set destination to 62.99.171.21, port 5060 Reliably Transmitting (NAT) to 62.99.171.21:5060: BYE sip:020316695199@62.99.171.52:5060 SIP/2.0 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK1e9c2747;rport Route: From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Contact: Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="0316890841", realm="voip.inode.at", algorithm=MD5, uri="sip:020316695199@62.99.171.52:5060", nonce="423ed71f8861d4f021fc69c6d9658dbe39545106", response="951f5f01f2ac70612a7af3873bda923c", opaque="" Content-Length: 0 --- == Spawn extension (default, 00695199, 1) exited non-zero on 'SIP/70-ee64' JoIP2000i*CLI> <-- SIP read from 62.99.171.21:5060: SIP/2.0 200 OK Call-ID: 00885e1b762721104c04a8af1716703b@81.223.57.205 CSeq: 104 BYE From: "70" ;tag=as100f59dc To: ;tag=de56240ac355625 Via: SIP/2.0/UDP 81.223.57.205:5060;branch=z9hG4bK1e9c2747;rport=5060 Record-Route: Content-Length: 0 Supported: replaces User-Agent: SN2400 MxSF v3.2.7.36 --- (10 headers 0 lines)--- Response message is BYE Destroying call '00885e1b762721104c04a8af1716703b@81.223.57.205'