Asterisk CVS-HEAD-01/06/05-00:46:46, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-01/06/05-00:46:46 currently running on voip (pid = 13600) Verbosity is at least 5 Asterisk Ready. -- Remote UNIX connection voip*CLI> sip debug SIP Debugging Enabled voip*CLI> Sip read: INVITE sip:112@192.168.0.100;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK75b1b981 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 248 Accept: application/sdp v=0 o=Cisco-SIPUA 6463 14531 IN IP4 192.168.0.101 s=SIP Call c=IN IP4 192.168.0.101 t=0 0 m=audio 18134 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.101 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.101:18134 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK75b1b981 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as2db35a83 Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 101 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="15b950a6" Content-Length: 0 to 192.168.0.101:5060 Scheduling destruction of call '001192d9-83070014-6241a101-08f783a2@192.168.0.101' in 15000 ms Found user '1' voip*CLI> Sip read: ACK sip:112@192.168.0.100;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK75b1b981 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as2db35a83 Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 101 ACK Content-Length: 0 7 headers, 0 lines voip*CLI> Sip read: INVITE sip:112@192.168.0.100;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK1dd57c63 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Proxy-Authorization: Digest username="1",realm="asterisk",uri="sip:192.168.0.100",response="f9681497ccadc420ee309d8340fbb672",nonce="15b950a6",algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 248 v=0 o=Cisco-SIPUA 6463 14531 IN IP4 192.168.0.101 s=SIP Call c=IN IP4 192.168.0.101 t=0 0 m=audio 18134 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.101 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.101:18134 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found user '1' Looking for 112 in sip list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK1dd57c63 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 102 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.101:5060 -- Executing Dial("SIP/1-e594", "sip/612@fwd.pulver.com|60") in new stack We're at 129.39.222.183 port 18408 Answering/Requesting with root capability 8 Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:612@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 129.39.222.183:5060;branch=z9hG4bK69679d8d From: "Interno 1" ;tag=as5564b61b To: Contact: Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 102 INVITE User-Agent: "ChersoVoip" Date: Fri, 07 Jan 2005 13:44:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 13648 13648 IN IP4 129.39.222.183 s=session c=IN IP4 129.39.222.183 t=0 0 m=audio 18408 RTP/AVP 8 3 0 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 69.90.155.70:5060 -- Called 612@fwd.pulver.com voip*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 129.39.222.183:5060;branch=z9hG4bK69679d8d;received=82.50.xxx.xxx From: "Interno 1" ;tag=as5564b61b To: Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 102 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 8 headers, 0 lines voip*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 129.39.222.183:5060;received=82.50.xxx.xxx;branch=z9hG4bK69679d8d From: "Interno 1" ;tag=as5564b61b To: ;tag=as0e9103b2 Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 10 headers, 0 lines -- SIP/fwd.pulver.com-25a8 is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK1dd57c63 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 102 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.101:5060 voip*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 129.39.222.183:5060;received=82.50.xxx.xxx;branch=z9hG4bK69679d8d Record-Route: From: "Interno 1" ;tag=as5564b61b To: ;tag=as0e9103b2 Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 5772 5772 IN IP4 69.90.168.13 s=session c=IN IP4 69.90.168.13 t=0 0 m=audio 12816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 12 headers, 10 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 69.90.168.13:12816 Found description format PCMU Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.70, port 5060 Transmitting: ACK sip:612@69.90.168.13:5028 SIP/2.0 Via: SIP/2.0/UDP 129.39.222.183:5060;branch=z9hG4bK55a71ecd Route: From: "Interno 1" ;tag=as5564b61b To: ;tag=as0e9103b2 Contact: Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 102 ACK User-Agent: "ChersoVoip" Content-Length: 0 (no NAT) to 69.90.155.70:5060 -- SIP/fwd.pulver.com-25a8 answered SIP/1-e594 We're at 192.168.0.100 port 18902 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK1dd57c63 From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 102 INVITE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 268 v=0 o=root 13648 13648 IN IP4 192.168.0.100 s=session c=IN IP4 192.168.0.100 t=0 0 m=audio 18902 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.101:5060 -- Attempting native bridge of SIP/1-e594 and SIP/fwd.pulver.com-25a8 voip*CLI> Sip read: ACK sip:112@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK7e7d9d1c From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 102 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="1",realm="asterisk",uri="sip:192.168.0.100",response="f9681497ccadc420ee309d8340fbb672",nonce="15b950a6",algorithm=md5 Content-Length: 0 9 headers, 0 lines voip*CLI> Sip read: BYE sip:112@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK5a95eacd From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 103 BYE User-Agent: CSCO/7 Content-Length: 0 Proxy-Authorization: Digest username="1",realm="asterisk",uri="sip:192.168.0.100",response="3054ffc71e08d5ec7c5c1c44d5a9da0f",nonce="15b950a6",algorithm=md5 9 headers, 0 lines Sending to 192.168.0.101 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK5a95eacd From: "Interno 1" ;tag=001192d9830700174428fa08-2783fe01 To: ;tag=as18852c3d Call-ID: 001192d9-83070014-6241a101-08f783a2@192.168.0.101 CSeq: 103 BYE User-Agent: "ChersoVoip" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 voip*CLI> to 192.168.0.101:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.70, port 5060 Reliably Transmitting: BYE sip:612@69.90.168.13:5028 SIP/2.0 Via: SIP/2.0/UDP 129.39.222.183:5060;branch=z9hG4bK2c466110 Route: From: "Interno 1" ;tag=as5564b61b To: ;tag=as0e9103b2 Contact: Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 103 BYE User-Agent: "ChersoVoip" Content-Length: 0 (no NAT) to 69.90.155.70:5060 == Spawn extension (sip, 112, 1) exited non-zero on 'SIP/1-e594' voip*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 129.39.222.183:5060;received=82.50.xxx.xxx;branch=z9hG4bK2c466110 Record-Route: From: "Interno 1" ;tag=as5564b61b To: ;tag=as0e9103b2 Call-ID: 7e3c1b0024556fbc731648d2018cee65@129.39.222.183 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 11 headers, 0 lines Destroying call '7e3c1b0024556fbc731648d2018cee65@129.39.222.183' Destroying call '001192d9-83070014-6241a101-08f783a2@192.168.0.101'